Re: [pulseaudio-discuss] 'Failed to find a working profile' for firewire sound devices
Hi David, AFAIK, the driver should support what the hardware supports. Yes. It's my intent to add the rules between channels/rates. Well, today I did some tests for an idea to use 'plug' plugin. 1) Edit /usr/share/pulseaudio/alsa-mixer/profile-sets/default.conf and change hw:%f to plughw:%f in the section Mapping analog-stereo. If the plughw plugin can upscale, then that should make things work. This wasn't good. Then I remembered this fact: Hm. Actually PulseAudio tries opening plug:hw:1 too and setting the channel number to two fails in that case too. If the plug plugin can change the number of channels, then that should have been working. Then I debug PulseAudio/libasound in a point of 'how to open ALSA PCM handle'. I have an presumption that 'plug' plugin can upscale channels correctly if PulseAudio open 'plug:hw' device as the same way as normal-aplay open. As a result, I got a conviction that 'SND_PCM_NO_AUTO_CHANNELS' flag for snd_pcm_open() disables upscaling. I can reproduce it with this program (it's too rough): http://ubuntuone.com/4T5tqqiioUZMFBe4d5I4Zn So I conclude we cannot use 'plug' plugin for this purpose. Would you please confirm this in your side? And for these two options below, I need more time to test. To aid for applications, one can add custom per-card mapping in alsa-lib - e g, look at /usr/share/alsa/cards/ICE1712.conf where you can open the device front:card with two channels and it automatically upscales to 10 or 12 channels. 2) Alternatively, you could add channels to your mapping: channel-map = left,right,aux0,aux1,aux2,aux3,aux4,aux5,aux6,aux7, aux8,aux9,aux10,aux11 Thanks for your help Takashi Sakamoto o-taka...@sakamocchi.jp ___ pulseaudio-discuss mailing list pulseaudio-discuss@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] raop no sound
I am running a somewhat oldish opensuse installation. Since recently there is an apple airport available on the network, with wired access (it is configured to offer wlan to guests) I wanted to try that and so changed the pulseaudio from .9.12 to .9.15 and built the missing module (which suse does not ship normally) I can load the module pactl load-module module-raop-sink server=192.168.2.31 and some network activity is going on. The output appears in pavucontrol I tried to follow the network exchanges, they seem to make sense and follow documents I have found on the net, with one difference: I could net see crypto stuff coming back from the device actually ANNOUNCE, SETUP, RECORD, and SET_PARAMETER are sent in this order and all get an OK response. At that time pulseaudio decides that there is currently no sound, and breaks the connection. When I start playing sound a moment later, nothing happens. Is this a problem related to the early version (in that case: is it possible to upgrade PA without reinstalling most of the system ... e.g. dependencies on alsa version) or is the raop module simply incompatible with airports that ship today Regards Wolfgang ___ pulseaudio-discuss mailing list pulseaudio-discuss@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] 'Failed to find a working profile' for firewire sound devices
Hi Alexander, This is documented at http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#ga64fa40b556374dabe40d4874242fee19 By forced channel conversion they mean the route ALSA plugin. See how it is done in /usr/share/alsa/cards/ICE1712.conf , which is similar to your use case. Specifically, they try to upscale everything to 10 channels, because this is what this hardware supports directly. Yes, I know. I have no conviction whether it actually effects 'plug' plugin or not. And I don't know PulseAudio always uses it when doing snd_pcm_open(). By forced channel conversion they mean the route ALSA plugin. See how it is done in /usr/share/alsa/cards/ICE1712.conf , which is similar to your use case. Specifically, they try to upscale everything to 10 channels, because this is what this hardware supports directly. Here I have a question because I know little about the devices. (I should post this question to alsa-devel but this is good opportunity for me to get information from you.) My drivers, 'snd-fireworks/snd-bebob/snd-oxfw' totally supports 80-90 devices. Each device has different combination between channels/rates. Can I write configuration to cover such variation? For example, let me assume there are four models which 'snd-bebob' can support: (regend: 44.1/48.0/88.2/96.0/176.4/192.0) For playback: Model_A: 2ch/2ch/2ch/2ch/2ch/2ch Model_B: 12ch/12ch/10ch/10ch/4ch/4ch Model_C: 16ch/16ch/8ch/6ch/4ch/2ch Model_D: 34ch/34ch/34ch/34ch/20ch/20ch For capture: Model_A: 2ch/2ch/2ch/2ch/2ch/2ch Model_B: 8ch/8ch/6ch/6ch/2ch/2ch Model_C: 34ch/34ch/34ch/18ch/18ch/18ch Model_D: 40ch/40ch/30ch/30ch/20ch/20ch And 'Model_D' changes its channel formation according to internal setting: For playback: Model_d: 12ch/12ch/12ch/12ch/10ch/10ch For capture: Model_d: 20ch/20ch/15ch/15ch/10ch/10ch Can I write a configuration file to cover them for 'pcm.front'? Thanks Takashi Sakamoto o-taka...@sakamocchi.jp ___ pulseaudio-discuss mailing list pulseaudio-discuss@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] 'Failed to find a working profile' for firewire sound devices
Hi Alexander, AFAIK PulseAudio always uses this flag. I've made it clear in my previous mail. No. I have deleted the table from my reply, but it looks like there are two issues here. First, the needed conversions depend on the sample rate. Second, they vary between models and depend on hardware switches. I agree with your analysis for this issue. Is it really true that your hardware accepts two playback channels at 192 kHz, but has no way, at the register level, to accept two playback channels at lower rates? I cannot understand what you asked. Would you explain your intent for this question? I have already explain about relationships between channels/rate which the devices supports. For the second problem, I think, it is possible (but not necessarily a good idea) to have per-model and maybe per-switch-position configuration files. See how it is done in the cmipci driver (grep for SWIEC, look for the alternative configuration file with SWIEC in the name). I'm a bit negative for this idea when considering about how much work it takes for me. If the number of devices which my drivers support was 20 or less, I would have consider about it. But they support much devices than you expected. Regards Takashi Sakamoto o-taka...@sakamocchi.jp ___ pulseaudio-discuss mailing list pulseaudio-discuss@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] raop no sound
Hi Wolfgang, What Airport device are you using? Perhaps your device only supports raop2 protocol, which is not supported by the latest stable pulseaudio. I and some collaborators are now working on raop2 support. You may find this interesting. http://hfujita.github.io/pulseaudio-raop2/ Thanks, Hajime Hi Hajime, this is beginning of talk - I would believe it is understanding but I wanted to figure out first.. At least it seems to accept the TCP suggestion (and it is taking in data via tcp) ANNOUNCE rtsp://192.168.2.20/322896 RTSP/1.0 CSeq: 1 Content-Type: application/sdp Content-Length: 571 User-Agent: iTunes/4.6 (Macintosh; U; PPC Mac OS X 10.3) Client-Instance: Apple-Challenge: . v=0 o=iTunes 322896 IN INP4 192.168.2.20 s=iTunes c=IN IP4 192.168.2.31 t=0 0 m=audio 0 RTP/AVP 96 a=rtpmap:96 Apple Lossless a=fmtp:96 4096 0 16 40 10 14 2 255 0 0 44100 a=rsaaeskey:H4. a=aesiv: RTSP/1.0 200 OK Server: AirTunes/105.1 CSeq:1 SETUP rtsp://192.168.2.20/322896 RTSP/1.0 CSeq: 2 Transport: RTP/AVP/TCP;unicast;interleaved=0-1;mode=record User-Agent: iTunes/4.6 (Macintosh; U; PPC Mac OS X 10.3) Client-Instance: RTSP/1.0 200 OK Transport: RTP/AVP/TCP;unicast;mode=record;server_port=6000 Session: 1 Audio-Jack-Status: connteced; type=analog Server: AirTunes/105.1 CSeq: 2 ___ pulseaudio-discuss mailing list pulseaudio-discuss@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] raop no sound
Hmm, you are right. Your device seems to understand the TCP version. I suppose the current stable pulseaudio should support your device. Unfortunately I don't have any experience with TCP version of the protocol, so I don't have a clue at this moment. A few things that came up to my mind... - usually module-raop-discover will automatically find the device, so you don't have to manually load module-raop-sink. Have you installed some additional modules like pulseaudio-zeroconf or something? - If you have paprefs command in your system, launch it and enable Airtunes from the GUI interface there. - Attaching a log would help us understanding the issue more precisely. Thanks, Hajime haman...@t-online.de wrote: Hi Wolfgang, What Airport device are you using? Perhaps your device only supports raop2 protocol, which is not supported by the latest stable pulseaudio. I and some collaborators are now working on raop2 support. You may find this interesting. http://hfujita.github.io/pulseaudio-raop2/ Thanks, Hajime Hi Hajime, this is beginning of talk - I would believe it is understanding but I wanted to figure out first.. At least it seems to accept the TCP suggestion (and it is taking in data via tcp) ANNOUNCE rtsp://192.168.2.20/322896 RTSP/1.0 CSeq: 1 Content-Type: application/sdp Content-Length: 571 User-Agent: iTunes/4.6 (Macintosh; U; PPC Mac OS X 10.3) Client-Instance: Apple-Challenge: . v=0 o=iTunes 322896 IN INP4 192.168.2.20 s=iTunes c=IN IP4 192.168.2.31 t=0 0 m=audio 0 RTP/AVP 96 a=rtpmap:96 Apple Lossless a=fmtp:96 4096 0 16 40 10 14 2 255 0 0 44100 a=rsaaeskey:H4. a=aesiv: RTSP/1.0 200 OK Server: AirTunes/105.1 CSeq:1 SETUP rtsp://192.168.2.20/322896 RTSP/1.0 CSeq: 2 Transport: RTP/AVP/TCP;unicast;interleaved=0-1;mode=record User-Agent: iTunes/4.6 (Macintosh; U; PPC Mac OS X 10.3) Client-Instance: RTSP/1.0 200 OK Transport: RTP/AVP/TCP;unicast;mode=record;server_port=6000 Session: 1 Audio-Jack-Status: connteced; type=analog Server: AirTunes/105.1 CSeq: 2 ___ pulseaudio-discuss mailing list pulseaudio-discuss@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
